Method and apparatus for the control of multimedia services in networks

ABSTRACT

A method, apparatus, and system for reducing congestion of real time data traffic on a multimedia communications network having a traffic control mechanism. Information regarding congestion of the multimedia communications network is first extracted from data traffic in the multimedia communications network. This extraction is performed by a network of monitors. Congestion is regulated by a central server, which receives network information from the monitors and utilizes the network information to analyze congestion status and communicate instructions to the multimedia communications network to reduce congestion.

RELATED PATENT APPLICATIONS

The present application is a continuation of U.S. patent applicationSer. No.: 09/212,650, titled “Method for the Control of MultimediaServices in Networks,” filed on Dec. 16, 1998, now U.S. Pat. No.6,529,475.

BACKGROUND OF THE INVENTION

1. Technical Field

The present invention relates generally to a communications system andin particular to a method, apparatus, and system for improving flow ofdata traffic within a communications network. Still more particularly,the present invention relates to a method and system for improving flowof data traffic within a multimedia communications network by reducingcongestion.

2. Description of the Related Art

The H.323 standard is an umbrella recommendation from the InternationalTelecommunication Union (ITU) that sets standards for multimediacommunications over Local Area Networks (LANs) that do not provide aguaranteed Quality of Service (QoS). These networks dominate today'scorporate desktops and include packet-switched Transmission ControlProtocol/Internet Protocol (TCP/IP) and Internet Packet Exchange (IPX)over Ethernet, Fast Ethernet and Token Ring network technologies.Therefore, the H.323 standard is an important building block for a broadnew range of collaborative, LAN-based applications for multimediacommunications.

The H.323 standard is the newest member of a family of ITU umbrellarecommendations which cover video telephone multimedia communicationsover a variety of pipelines. The H.323 standard is in many senses aderivative of H.320, 1990 umbrella recommendation for video telephoneover switched digital telephone networks. The H.323 standard borrowsheavily from H.320's structure, modularity, and audio/video compression/decompression (codec) recommendations.

The H.323 standard provides a foundation for audio, video, and datacommunications across IP based networks, including the Internet. Bycomplying to the H.323 standard, multimedia products and applications,from multiple vendors can interoperate, allowing users to communicatewithout concern for compatibility. The H.323 standard will be thekeystone for LAN based products for consumer, business, entertainment,and professional applications.

Communications under the H.323 standard can be considered a mix ofaudio, video, and control signals. Audio capabilities, Q.931 call setup,RAS control, and H.245 signaling are required. All other capabilitiesincluding video and data conferencing are optional. When multiplealgorithms are possible, the algorithm utilized by the encoder arederived from information passed by the decoder during the H.245capability exchange. H.323 terminals are also capable of asymmetricoperation (different encode and decode algorithm) and can send/receivemore than one video and audio channel.

The H.323 standard addresses call control, multimedia management, andbandwidth management for point-to-point and multipoint conferences. Itis designed to run on common network architectures. As networktechnology evolves, and as bandwidth management techniques improve,H.323-based solutions will be able to take advantage of the enhancedcapabilities. The H.323 standard is not tied to any hardware oroperating system and H.323-compliant platforms will be available in allsizes and shapes, including video-enabled personal computers, dedicatedplatforms, and turnkey boxes.

Often, users want to conference without worrying about compatibility atthe receiving point. The H.323 standard establishes standards forcompression and decompression of audio and video data streams, ensuringthat equipment from different vendors will have some area of commonsupport. Besides ensuring the receiver can decompress the information,the H.323 standard establishes methods for receiving clients tocommunicate capabilities to the sender. The standard also establishescommon call setup and control protocols.

The H.323 standard utilizes both reliable and unreliable iscommunications. Control signals and data require reliable transportbecause the signal must be received in the order in which they were sentand cannot be listed. Audio and video streams lose their value withtime. If a packet is delayed, it may not have relevance to the end user.Audio and video signals utilize the more efficient but less reliabletransport.

Because the H.323 standard is Real-Time Transport Protocol (RTP) based,it can operate on the Internet's Multicast Backbone (Mbone), a virtualnetwork on top of the Internet that provides a multicast facility, andsupports video, voice and data conferencing. The H.323 [H.323v2]standard has been proposed to perform call control (i.e. makeconnections) of real-time service on IP networks. The H.323 standardallows end-points or terminals wanting to make connections to negotiatebandwidth and coding requirements before the connection is established.In this standard there are three key players:

-   -   End-point: These are terminals which need to make connections.        They request the connection through a gatekeeper (if one is on        the network) and they also negotiate the connection parameters.    -   Gatekeeper: These entities perform bandwidth control (on LANs)        and routing of connection packets towards the destination        terminal.    -   Gateway: This entity can be thought of as a collection of        end-points, but these entities also translate from other bearer        protocols (such as time-division multiplexing (TDM)) to the IP        protocol.

FIG. 1 clearly illustrates the interconnectivity of these components.FIG. 1 depicts a network with several gatekeepers 100, routers 101,endpoints 102, gateways 108, and terminals 104. Gatekeepers 100, routers101, gateways 108, endpoints 102, and terminals 104 are interconnectedvia network links 106. Note that gatekeepers 100 are linked together toform the framework of the network while gateways 108, endpoints 102 andterminals 104 serve as the branches to this framework.

The Gatekeeper is a H.323 entity that provides address translation,control access, and sometimes bandwidth management to the LAN for H.323terminals, Gateways, and Multipoint Control Units (MCUs). Gatekeepersperform two important call control functions which help preserve theintegrity of the corporate data network. The first is addresstranslation from LAN aliases for terminals and gateways to IPXaddresses, as defined in the Registration/Admission/Status (RAS)specification. The second function is bandwidth management, which isalso designated within RAS. For instance, if a network manager hasspecified a threshold for the number of simultaneous conferences on theLAN, the Gatekeeper can refuse to make any more connections oncethreshold is reached. The effect is to limit the total conferencingbandwidth to some fraction of the total available, the remainingcapacity is left for e-mail, file transfers, and other LAN protocols.The collection of all Terminals, Gateways and Multipoint Control Unitsmanaged by a single gatekeeper is known as a H.323Zone.

Improvements in communications arise from changing user's needs anddemands. Previously, public network needs were driven by telephoning andvoice data. Data traffic has grown slowly until recently. With the lowercost in telecommunications and the higher increase in processing powerof computers, the number of users accessing communications networks hasincreased. The needs of these users include, for example, videotelephone, low cost video conferencing, imaging, high definitiontelevision (HDTV), and other applications requiring multimedia datatransfers. Multimedia combines different forms of media in thecommunication of information between a user and a data processingsystem, such as a personal computer. A multimedia application is anapplication that utilizes different forms of communications within asingle application. Multimedia applications may, for example,communicate data to a user on a computer via audio, text, and videosimultaneously. Such multimedia applications are usually bit intensive,real time, and very demanding on communications networks.

The H.323 standard sets multimedia standards for the existinginfrastructure (i.e. IP-based networks). Design to compensate for theeffect of highly variable LAN latency, the H.323 standard allowscustomers to utilize multimedia applications without changing theirnetwork infrastructure.

Reliable transmission of messages utilizes a connection-oriented modefor data transmission. Reliable transmission guarantees sequencederror-free, flow-controlled transmission of packets, but can delaytransmission and reduce throughput. The H.323 standard utilizes reliable(TCP) end-to-end service for the H.245 Control Channel, the T.120 DataChannel and the Call Signaling Channel.

Within the IP stack, unreliable services are provided by User DatagramProtocol (UDP). Unreliable transmission is a mode without connectionwith promises nothing more than “best effort” delivery. UDP offersminimal control information. It is a network layer which sits at thesame level of network stack as TCP. It is a connection-less protocolwithin TCP/IP that corresponds to the transport layer in the ISO/OSImodel. UDP converts data messages generated by an application intopackets to be sent via IP but does not verify that messages have beendelivered correctly. The H.323 standard utilizes UDP for the audio,video and the RAS Channel.

IP networks are the technology driving the Internet. The rise of thesenetworks is primarily due to their acceptance as the layer 3 protocol inthe enterprise networks. Most PCs now utilize transmission controlprotocol/Internet protocol (TCP/IP) as their networking protocol. IP haseven gained acceptance as the wide area protocol since it is about25-30% more efficient than ATM.

The kinds of traffic running over IP networks are of two major types:

-   -   Elastic traffic or non-real-traffic which is primarily data file        transfer. Most of this traffic uses TCP as its transport level        protocol and it can withstand delay quite well, but any        corruption of data must be retransmitted; and    -   The inelastic or real-time traffic is interactive voice, video        or data-conferencing. This kind of traffic does not withstand        delay well since late information in an interactive session is        of no use. This kind of traffic utilizes real time protocol        (RTP) over UDP as the transport protocol.

UDP is the dominant multimedia protocol. However, it does not have anyinherent congestion control mechanism. A need thus exists for Real-TimeTransport Control Protocol (RTCP) protocol on top of UDP to controldelays. RTCP works with RTP for multimedia services.

Running real-time traffic over IP network has other significant problemsalso. Currently, there is no way of reserving bandwidth end-to-end in anIP network. Each IP packet takes its own route through the network.Therefore, each packet gets to its destination (in theory) through adifferent route and can have a different delay in getting to itsdestination. This causes delay variance or jitter at the destinationwhere the packets have to be “played” for the destination user.

There have been some concerns in voice-over-IP (VoIP) industry that theintroduction of large volume voice traffic into an IP network willunfairly compete for network bandwidth with existing TCP traffic. TCPhas congestion control mechanisms built in. Once TCP senses networkcongestion by its detection of lost packets, it will reduce its packettransmission rate. Therefore, in case of network congestion, all TCPconnections will throttle back until the congestion is relieved.However, UDP does not have similar control mechanisms. For now, UDPtraffic in IP networks has been minimal. Although only TCP trafficreacts to network congestion, it has not been a problem. It is expectedthat the introduction of VoIP services will bring in a large volume ofUPD traffic. Voice UDP traffic is an ill-behaved source and canpotentially lock out TCP traffic in case of congestion. Since othermulti-media services, such as video conferencing, are also expected touse UDP as the transport layer protocol, this problem exits for all IPmulti-media services. Data applications currently use and will continueto use reliable transmission protocols (i.e. TCP) because data integrityis the top priority. The perceived UDP traffic increase will come fromIP multimedia applications. Some congestion control mechanism isrequired to manage multi-media UDP traffic.

Therefore, it would be desirable to have an improved method andapparatus for reducing congestion in the flow of data traffic in amultimedia communications network. Additionally, it would be desirableto reduce such congestion flow without significant interruption in theflow of data within the multimedia communications network.

SUMMARY OF THE INVENTION

It is one object of the present invention to provide an improved methodand system for a communications system.

It is another object of the present invention to provide an improvedmethod and system for improving flow of data traffic within acommunications network.

It is yet another object of the present invention to provide an improvedmethod and system for improving flow of data traffice within amultimedia communications network by reducing congestion.

The above features are achieved as follows. A method is disclosed forreducing congestion of real time data traffic on a multimediacommunications network having a traffic control mechanism. The methodcomprises first extracting from data traffic in the multimediacommunications network information regarding congestion of themultimedia communications network. Secondly, congestion is regulated onthe multimedia communications network utilizing the network informationextracted from the multimedia communications network.

In accordance with a preferred embodiment of the present invention, aplurality of monitors scans the through data traffic for RTCP packets.The RTCP packets provide information on the traffic flow which isextracted by the monitors. The information is forwarded to a centralserver where it is analyzed. Following this analysis, the central serverinitiates steps to relieve congestion in the network.

The above as well as additional objectives, features, and advantages ofthe present invention will become apparent in the following detailedwritten description.

BRIEF DESCRIPTION OF THE DRAWINGS

The novel features believed characteristic of the invention are setforth in the appended claims. The invention itself, however, as well asa preferred mode of use, further objectives and advantages thereof, willbest be understood by reference to the following detailed description ofan illustrative embodiment when read in conjunction with theaccompanying drawings, wherein:

FIG. 1 is a diagram depicting a representation of physical network of acommunications system according to one embodiment of the presentinvention;

FIG. 2 is a diagram of a communications system with attached congestionmonitors according to one illustration of the present invention; and

FIG. 3 is a flow chart depicting the logic flow of the process forrelieving congestion in the communications system according to oneembodiment of the present invention.

DETAILED DESCRIPTION

With reference now to the Figures, the preferred embodiment of thepresent invention is depicted. The invention discloses a way to monitorcongestion of data traffic on a multimedia network. While the preferredembodiment of the invention described herein is implemented within aH.323 network, it is understood by those skilled in the art that theinvention can be practiced in any network containing the RTCP protocolor any other protocol with similar functionality.

Video and audio traffic is bandwidth intensive and could clog thecorporate network. The H.323 standard addresses this issue by providingbandwidth management. Network managers can limit the number ofsimultaneous H.323 connections within their network or limit the amountof bandwidth available to H.323 applications. These limits ensure thatcritical traffic will not be disrupted.

In RTP protocol, a header containing a time stamp and a sequence numberis added to each UDP packet. With appropriate buffering at the receivingstation, timing and sequence information allows the application toeliminate duplicate packets, reorder out of sequence packets,synchronize sound, video and data, and accept continuous playback inspite of varying latencies.

The Real Time Control Protocol (RTCP) is utilized for the control ofRTP. RTCP monitors the quality of service, conveys information about thesession participants, and periodically distributes control packetscontaining quality information to all session participants through thesame distribution mechanisms as the data packets.

The network protocol stack for multi-media services specified in theH.323 standard is one that puts application traffic on top of RTP/RTCP,then on top of UDP/IP. Within this protocol stack, RTCP does providefeedback information on the quality of the data distribution. RTCP is ascalable protocol that provides sufficient network delay, packet loss,throughput information, etc. In the present invention, the RTCPinformation is utilized to build congestion control mechanism formulti-media UDP traffic and multi-media traffic in general becauseRTP/RTCP is used for all multi-media traffic.

According to the H.323 standard, voice traffic (video traffic as well)is transmitted on top of RTP/RTCP. The real-time transport protocol(RTP) provides end-to-end delivery services for data with real-timecharacteristics, such as interactive audio and video. These servicesinclude payload type identification, sequence numbering, time-stampingand delivery monitoring. The RTP control protocol (RTCP) is based on theperiodic transmission of control packets to all participants in thesession. It provides feedback on the quality of the data distribution,carries a persistent transport-level identifier for RTP sources, andcontrols the RTCP packet rate to scale up to a large number ofparticipants. RTCP also dynamically keeps track of the number ofparticipants in a session and guarantees the control traffic is limitedto a small fraction of the session bandwidth (suggested at 5%).

RTCP information is not only useful for the sender and receiver but alsocan be utilized by third-party-monitors. Such monitors (congestionmonitors) process only the RTCP control packets and not thecorresponding RTP data packets to evaluate the performance of networks.Congestion monitors can be deployed to monitor RTCP messages (possiblysome ICMP messages as well, such as the source quench message) andderive congestion information about the network. Such information isutilized by the call control servers, gatekeepers, in the network tocontrol the network congestion and provide QoS guarantees.

The present invention provides a congestion control mechanism for IPmulti-media services based on H.323 networks and other networks whichcontain the RTCP protocol or protocols with similar functionality.Basically, congestion monitors are utilized to monitor RTCP informationgenerated by H.323 sessions and communicate with a central congestionserver to calculate the network congestion status. In the preferredembodiment, the congestion monitors are specialized stand-alone boxesthat can tap into a router or switch to monitor RTCP traffic. In anotherembodiment of the invention, the congestion monitor is integrated withinthe IP routers. Its primary functionality is to scan RTCP packets withinthe traffic stream, extract performance information from the packets,and summarize them on session basis. Some simple performance statisticscan be calculated at the monitors. Call admission control logic isimplemented at network call control centers, gatekeepers, to controlnetwork congestion.

The congestion monitors are specialized because they can look only atRTCP packets. Unlike routers, for example, which look at and processesall the data packets, congestion monitors can differentiate the packetsand scan only the RTCP packets within the data traffic. These RTCPpackets represent a small percentage of the overall data traffic. Thecongestion monitor may be hardware implemented. Due to its need to reactto the high speed of data traffic, congestion monitors themselves arerequired to have very high processing speed. The congestion monitorsperforms only basic processing of the packets to collect basicstatistical information; therefore they do not have to be very complex(little intelligence).

The appropriate usage of congestion monitors can reduce the delayvariance and packet drop rate of IP packets and, thus, improve Qualityof Service (QoS) for multi-media IP services. There are two possibleapproaches to implement control over network congestion. These are calladmission control and bandwidth reduction.

Call admission control is probably the more effective approach for voiceover IP services. It is also very effective for other type of services.In the preferred embodiment, call admission control (i.e., rejecting newcalls) is implemented at network call control centers, gatekeepers, tocontrol network congestion. Whenever the centralized congestion serverdetects congestion in the network, it will inform the relevantgatekeepers in the network. If the congestion is global, all gatekeeperswill be informed. If it is local to certain part of the network, onlygatekeepers controlling that part of the network will be informed in thepreferred embodiment. Upon such notification, gatekeepers will adjustits call admission policies according to certain predefined rules. Underthese rules in the preferred embodiment, it is tougher for a new call tobe admitted the more congested the network is. Also, the rules providefairness among users and enforce service priority if required. Since thecentral congestion server utilizes real time RTCP information generatedby RTP connections, the scheme can provide responsive and timely controlover voice multi-media UDP traffic in a IP network.

Bandwidth reduction is the second approach to controlling the networkcongestion. Unlike data applications, which are delay insensitive,multi-media applications require certain constrains on end-to-end datadelivery. Therefore, the schemes utilized in TCP protocol may introduceunconstrained delay. Bandwidth reduction can not be achieved byincreasing transmission time and, consequently, reducing thetransmission rate. Instead the amount of data that needs transmissionshould be reduced for multi-media services in case of networkcongestion. For video, bandwidth reduction can be achieved by adjustingwindow size, frame rate, video quality, color coding schemes, etc. Invoice-only connections, such bandwidth reduction options are limited.For example, the lowest coding rate for VoIP is 5.3 kbps which,including all the header overhead, has an effective rate of only 16kbps. There are ways to reduce it further, such as silence compressionand header compression.

Normally, applications can detect congestion by RTCP information theyreceive and are expected to take certain actions to reduce its data ratein case of congestion. However, it is possible that even though someconnections do not see obvious congestion signals, the overall networkis congested. In order to effectively control the congestion, thenetwork may want to make these applications reduce their data rate aswell. There are two possible ways of doing it. In the preferredembodiment, the gatekeepers inform these applications directly. Theother embodiment allows the congestion monitors to alter the content ofRTCP packets to “fool” the source applications into congestion detectionand bandwidth reduction.

In the preferred embodiment, the balance between the TCP congestioncontrol schemes and the proposed RTCP based control scheme is a criticalissue. Fairness needs to be preserved between the two schemes accordingto service priority and bandwidth availability. Neither scheme takesaway an unfair amount of bandwidth at the same priority level from theother when congestion happens. Since the TCP schemes are wellstandardized whereas RTCP does not specify how to act on the congestion,RTCP schemes are tuned to react in a manner comparable to TCP undercongestion in the preferred embodiment.

The central servers are high capacity computers which take multipleconnections and can perform network management/configurations. Most ofthe functionality of the server is completed utilizing softwarealgorithms. In one embodiment of the invention, these central serversexist as independent units from the network and send/relay messages tothe gatekeepers to control congestion. In another embodiment of theinvention, the gatekeepers themselves serve as the central servers. Thisembodiment is possible since gatekeepers are software driven. Therequired functionality of the central server can be programmed into thegatekeepers (i.e. the software modified) to enable them to manage theflow of data traffic based on the information received form themonitors.

Application Specific Integrated Circuits (ASIC) technology can beutilized to implement the functionality to complete the task inrealtime. Some simple performance statistics can be calculated at thecongestion monitors, then, statistics from the sessions that arereporting performance problems will be sent to a centralized server tocompile the overall picture about the network congestion status. RTCPpackets provide sufficient information to derive statistics such as thepacket loss rate, average payload size, connection throughput, anddetermine whether problems are local, regional or global. Because therequired functionality is limited and RTCP traffic is low volume innature, such a monitor does not need to be a high-capacity product.Therefore, the implementation is relatively easy and inexpensive.

With reference now to the figures, and in particular FIG. 2, the presentinvention may be implemented with the basic communications network asshown. FIG. 2 depicts the preferred embodiment of the present invention.The main elements of the present invention are shown overlaying thenetwork of FIG. 1. The network as described above consist of gatekeepers100, routers 101, gateways 108, endpoints 102, and terminals 104interconnected via network links 106. Connected to routers 101, aremonitors 110 which remove the RTCP packets from the network. Thesemonitors send the information received from the RTCP packets to centralserver 112 via some transport/connection means 111 such as a bus link orother communication channel. Central server 112 is in turn connected tothe network's gatekeepers 100 via server connections 113. Note that inthis preferred embodiment, central server operates separately fromgatekeepers.

Turning now to FIG. 3, there is depicted one embodiment of the logicflow of the process for reducing congestion in the network. The processbegins when the congestion monitors are connected to and begin tomonitor the network as illustrated in block 201. The congestion monitorslook at RTCP packets within the data traffic as shown in block 202. Oncethis is done, the RTCP packets are analyzed in real time and statisticalinformation regarding the congestion status of the network are collectedas depicted in block 203. Periodically, the congestion monitors forwardthe information to the central server as shown in block 204. Centralserver utilizes this information to determine the congestion status ofthe network as shown in block 205. If the network is not congested, theprocess is restarted with the next batch of RTCP packets. If however,the network is congested, the central server sends a signal to thegatekeepers as illustrated in block 206. Once this signal is sent, thegatekeepers react to reduce congestion in the network as shown in block207. This process is then repeated for the next set of RTCP packets.

The description of the preferred embodiment of the present invention hasbeen presented for purposes of illustration and description, but is notintended to be exhaustive or limit the invention in the form disclosed.For example, although central servers are depicted as separatecomponents from the gatekeepers, it is understood that the functionalityof the central servers may be incorporated into existing gatekeepers.Many modifications and variations will be apparent to those of ordinaryskill in the art. The embodiment was chosen and described in order tobest explain the principles of the invention and the practicalapplication to enable others of ordinary skill in the art to understandthe invention for various embodiments with various modifications as aresuited to the particular use contemplated.

1. A method of reducing congestion of real time data traffic on amultimedia communications network having a plurality of gatekeepers,said method comprising: extracting, from specialized packets within datatraffic in said multimedia communications network, information regardingcongestion of said multimedia communications network, wherein saidextracting step is completed by a plurality of monitors coupled to saidmultimedia communications network that are designed to recognize andintercept said specialized packets; analyzing said information extractedfrom each of said plurality of monitors with a centralized processingcomponent coupled to said plurality of monitors; regulating congestionon said multimedia communications network utilizing the analyzedinformation whereby said centralized processing component is utilized tocontrol data traffic by messaging said plurality of gatekeepers anddynamically adjusting call admission policies of said gatekeepers towhich said control message is sent wherein said call admissions policiesare adjusted based on pre-determined rules upon receipt of said controlmessage.
 2. The method of claim 1, wherein said analyzing step analyzesan aggregate of all received congestion information against knownquality of service and transmission parameters of said network.
 3. Themethod of claim 1, wherein said specialized data packets are members ofa scalable protocol that provides network traffic information, includingat least one of: sufficient network delay information, packet lossinformation, and throughput information.
 4. The method of claim 3,wherein said scalable protocol provides feedback on quality of datadistribution.
 5. The method of claim 3, wherein said scalable protocolcarries a persistent transport level identifier.
 6. The method of claim3, wherein said scalable protocol dynamically tracks the number ofparticipants in a session and controls a real-time packet rate to scalefor different numbers of participants.
 7. The method of claim 3, whereinsaid scalable protocol is the Real Time Control Protocol (RTCP) and saidspecialized packets are RTCP packets.
 8. A system for reducingcongestion of real time data traffic on a multimedia communicationsnetwork having a traffic control mechanism, said system comprising:means for extracting, from specialized packets within data traffic insaid multimedia communications network, information regarding congestionof said multimedia communications network, wherein said extracting stepis completed by a plurality of monitors coupled to said multimediacommunications network that are designed to recognize and intercept saidspecialized packets, wherein said plurality of monitors are standalonecongestion monitors that differentiate said specialized packets fromstandard data packets within network traffic and said congestionmonitors are designed to retrieve only said specialized packets fromsaid network traffic; means for analyzing said information extractedfrom each of said plurality of monitors with a centralized processingcomponent coupled to said plurality of monitors; means for regulatingcongestion on said multimedia communications network utilizing theanalyzed information whereby said centralized processing component isutilized to control data traffic by messaging a plurality ofgatekeepers, and when a reduction in data rate is desired from atransmitting application, altering, via said plurality of monitors, acontent of said specialized packets to force congestion detection andsubsequent bandwidth reduction for said transmitting application.
 9. Thesystem of claim 8, wherein said means for analyzing further comprisesmeans for aggregating and comparing received congestion informationagainst known quality of service and transmission parameters of saidnetwork.
 10. The system of claim 8, wherein said specialized datapackets are members of a scalable protocol that provides network trafficinformation, including at least one of: sufficient network delayinformation, packet loss information, and throughput information. 11.The system of claim 10, wherein said scalable protocol provides feedbackon quality of data distribution.
 12. The system of claim 10, whereinsaid scalable protocol carries a persistent transport level identifier.13. The system of claim 10, wherein said scalable protocol dynamicallytracks the number of participants in a session and controls a real-timepacket rate to scale for different numbers of participants.
 14. Thesystem of claim 10, wherein said scalable protocol is the Real TimeControl Protocol (RTCP) and said specialized packets are RTCP packets.15. A system for reducing congestion of real time data traffic on amultimedia communications network having a traffic control mechanism,said system comprising: means for extracting, from specialized packetswithin data traffic in said multimedia communications network,information regarding congestion of said multimedia communicationsnetwork, wherein said extracting step is completed by a plurality ofmonitors coupled to said multimedia communications network that aredesigned to recognize and intercept said specialized packets; means foranalyzing said information extracted from each of said plurality ofmonitors with a centralized processing component coupled to saidplurality of monitors; and means for regulating congestion on saidmultimedia communications network utilizing the analyzed informationwhereby said centralized processing component is utilized to controldata traffic by messaging said plurality of gatekeepers; and means fordynamically adjusting call admission policies of said gatekeepers towhich said control message is sent wherein said call admissions policiesare adjusted based on predetermined rules upon receipt of said controlmessage.
 16. The apparatus of claim 15, wherein said means for analyzingfurther comprises means for aggregating and comparing receivedcongestion information against known quality of service and transmissionparameters of said network.
 17. The apparatus of claim 15, wherein saidspecialized data packets are members of a scalable protocol thatprovides network traffic information, including at least one of:sufficient network delay information, packet loss information, andthroughput information.
 18. The apparatus of claim 17, wherein saidscalable protocol provides feedback on quality of data distribution. 19.The apparatus of claim 17, wherein said scalable protocol carries apersistent transport level identifier.
 20. The apparatus of claim 17,wherein said scalable protocol dynamically tracks the number ofparticipants in a session and controls a real-time packet rate to scalefor different numbers of participants.
 21. The method of claim 17,wherein said scalable protocol is the Real Time Control Protocol (RTCP)and said specialized packets are RTCP packets.
 22. An apparatus forreducing congestion of real time data traffic on a multimediacommunications network having a traffic control mechanism, saidapparatus comprising: means for analyzing congestion informationextracted from specialized packets within data traffic in said network,wherein said specialized packets are intercepted and retrieved by eachof a plurality of monitors that are coupled to a centralized processingcomponent and are designed to recognize and intercept said specializedpackets; means for regulating congestion on said multimediacommunications network utilizing the analyzed information to controldata traffic by messaging a selected set among a plurality ofgatekeepers; and means for dynamically adjusting call admission policiesof said gatekeepers to which said control message is sent wherein saidcall admissions policies are adjusted based on predetermined rules uponreceipt of said control message.
 23. The apparatus of claim 22, whereinsaid means for analyzing further comprises means for aggregating andcomparing received congestion information against known quality ofservice and transmission parameters of said network.
 24. The apparatusof claim 22, wherein said specialized data packets are members of ascalable protocol that provides network traffic information, includingat least one of: sufficient network delay information, packet lossinformation, and throughput information.
 25. The apparatus of claim 24,wherein said scalable protocol provides feedback on quality of datadistribution.
 26. The apparatus of claim 24, wherein said scalableprotocol carries a persistent transport level identifier.
 27. Theapparatus of claim 24, wherein said scalable protocol dynamically tracksthe number of participants in a session and controls a real-time packetrate to scale for different numbers of participants.
 28. The method ofclaim 24, wherein said scalable protocol is the Real Time ControlProtocol (RTCP) and said specialized packets are RTCP packets.
 29. Anapparatus for reducing congestion of real time data traffic on amultimedia communications network having a traffic control mechanism,said apparatus comprising: means for analyzing congestion informationextracted from specialized packets within data traffic in said network,wherein said specialized packets are intercepted and retrieved by eachof a plurality of monitors that are coupled to a centralized processingcomponent and are designed to recognize and intercept said specializedpackets; means for regulating congestion on said multimediacommunications network utilizing the analyzed information to controldata traffic by messaging a selected set among a plurality ofgatekeepers; and means, when a reduction in data rate is desired from atransmitting application, for altering, via said plurality of monitors,a content of Real Time Control Protocol (RTCP) packets to forcecongestion detection and subsequent bandwidth reduction for saidtransmitting application.